// Copyright 2013 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.

#include "media/audio/alsa/alsa_input.h"

#include <stddef.h>

#include "base/bind.h"
#include "base/location.h"
#include "base/logging.h"
#include "base/macros.h"
#include "base/single_thread_task_runner.h"
#include "base/threading/thread_task_runner_handle.h"
#include "media/audio/alsa/alsa_output.h"
#include "media/audio/alsa/alsa_util.h"
#include "media/audio/alsa/alsa_wrapper.h"
#include "media/audio/alsa/audio_manager_alsa.h"
#include "media/audio/audio_manager.h"

namespace media {

static const int kNumPacketsInRingBuffer = 3;

static const char kDefaultDevice1[] = "default";
static const char kDefaultDevice2[] = "plug:default";

const char AlsaPcmInputStream::kAutoSelectDevice[] = "";

AlsaPcmInputStream::AlsaPcmInputStream(AudioManagerBase* audio_manager,
    const std::string& device_name,
    const AudioParameters& params,
    AlsaWrapper* wrapper)
    : audio_manager_(audio_manager)
    , device_name_(device_name)
    , params_(params)
    , bytes_per_buffer_(params.frames_per_buffer() * (params.channels() * params.bits_per_sample()) / 8)
    , wrapper_(wrapper)
    , buffer_duration_(base::TimeDelta::FromMicroseconds(
          params.frames_per_buffer() * base::Time::kMicrosecondsPerSecond / static_cast<float>(params.sample_rate())))
    , callback_(NULL)
    , device_handle_(NULL)
    , mixer_handle_(NULL)
    , mixer_element_handle_(NULL)
    , read_callback_behind_schedule_(false)
    , audio_bus_(AudioBus::Create(params))
    , weak_factory_(this)
{
}

AlsaPcmInputStream::~AlsaPcmInputStream() { }

bool AlsaPcmInputStream::Open()
{
    if (device_handle_)
        return false; // Already open.

    snd_pcm_format_t pcm_format = alsa_util::BitsToFormat(
        params_.bits_per_sample());
    if (pcm_format == SND_PCM_FORMAT_UNKNOWN) {
        LOG(WARNING) << "Unsupported bits per sample: "
                     << params_.bits_per_sample();
        return false;
    }

    uint32_t latency_us = buffer_duration_.InMicroseconds() * kNumPacketsInRingBuffer;

    // Use the same minimum required latency as output.
    latency_us = std::max(latency_us, AlsaPcmOutputStream::kMinLatencyMicros);

    if (device_name_ == kAutoSelectDevice) {
        const char* device_names[] = { kDefaultDevice1, kDefaultDevice2 };
        for (size_t i = 0; i < arraysize(device_names); ++i) {
            device_handle_ = alsa_util::OpenCaptureDevice(
                wrapper_, device_names[i], params_.channels(),
                params_.sample_rate(), pcm_format, latency_us);

            if (device_handle_) {
                device_name_ = device_names[i];
                break;
            }
        }
    } else {
        device_handle_ = alsa_util::OpenCaptureDevice(wrapper_,
            device_name_.c_str(),
            params_.channels(),
            params_.sample_rate(),
            pcm_format, latency_us);
    }

    if (device_handle_) {
        audio_buffer_.reset(new uint8_t[bytes_per_buffer_]);

        // Open the microphone mixer.
        mixer_handle_ = alsa_util::OpenMixer(wrapper_, device_name_);
        if (mixer_handle_) {
            mixer_element_handle_ = alsa_util::LoadCaptureMixerElement(
                wrapper_, mixer_handle_);
        }
    }

    return device_handle_ != NULL;
}

void AlsaPcmInputStream::Start(AudioInputCallback* callback)
{
    DCHECK(!callback_ && callback);
    callback_ = callback;
    StartAgc();
    int error = wrapper_->PcmPrepare(device_handle_);
    if (error < 0) {
        HandleError("PcmPrepare", error);
    } else {
        error = wrapper_->PcmStart(device_handle_);
        if (error < 0)
            HandleError("PcmStart", error);
    }

    if (error < 0) {
        callback_ = NULL;
    } else {
        // We start reading data half |buffer_duration_| later than when the
        // buffer might have got filled, to accommodate some delays in the audio
        // driver. This could also give us a smooth read sequence going forward.
        base::TimeDelta delay = buffer_duration_ + buffer_duration_ / 2;
        next_read_time_ = base::TimeTicks::Now() + delay;
        base::ThreadTaskRunnerHandle::Get()->PostDelayedTask(
            FROM_HERE,
            base::Bind(&AlsaPcmInputStream::ReadAudio, weak_factory_.GetWeakPtr()),
            delay);
    }
}

bool AlsaPcmInputStream::Recover(int original_error)
{
    int error = wrapper_->PcmRecover(device_handle_, original_error, 1);
    if (error < 0) {
        // Docs say snd_pcm_recover returns the original error if it is not one
        // of the recoverable ones, so this log message will probably contain the
        // same error twice.
        LOG(WARNING) << "Unable to recover from \""
                     << wrapper_->StrError(original_error) << "\": "
                     << wrapper_->StrError(error);
        return false;
    }

    if (original_error == -EPIPE) { // Buffer underrun/overrun.
        // For capture streams we have to repeat the explicit start() to get
        // data flowing again.
        error = wrapper_->PcmStart(device_handle_);
        if (error < 0) {
            HandleError("PcmStart", error);
            return false;
        }
    }

    return true;
}

snd_pcm_sframes_t AlsaPcmInputStream::GetCurrentDelay()
{
    snd_pcm_sframes_t delay = -1;

    int error = wrapper_->PcmDelay(device_handle_, &delay);
    if (error < 0)
        Recover(error);

    // snd_pcm_delay() may not work in the beginning of the stream. In this case
    // return delay of data we know currently is in the ALSA's buffer.
    if (delay < 0)
        delay = wrapper_->PcmAvailUpdate(device_handle_);

    return delay;
}

void AlsaPcmInputStream::ReadAudio()
{
    DCHECK(callback_);

    snd_pcm_sframes_t frames = wrapper_->PcmAvailUpdate(device_handle_);
    if (frames < 0) { // Potentially recoverable error?
        LOG(WARNING) << "PcmAvailUpdate(): " << wrapper_->StrError(frames);
        Recover(frames);
    }

    if (frames < params_.frames_per_buffer()) {
        // Not enough data yet or error happened. In both cases wait for a very
        // small duration before checking again.
        // Even Though read callback was behind schedule, there is no data, so
        // reset the next_read_time_.
        if (read_callback_behind_schedule_) {
            next_read_time_ = base::TimeTicks::Now();
            read_callback_behind_schedule_ = false;
        }

        base::TimeDelta next_check_time = buffer_duration_ / 2;
        base::ThreadTaskRunnerHandle::Get()->PostDelayedTask(
            FROM_HERE,
            base::Bind(&AlsaPcmInputStream::ReadAudio, weak_factory_.GetWeakPtr()),
            next_check_time);
        return;
    }

    int num_buffers = frames / params_.frames_per_buffer();
    uint32_t hardware_delay_bytes = static_cast<uint32_t>(GetCurrentDelay() * params_.GetBytesPerFrame());
    double normalized_volume = 0.0;

    // Update the AGC volume level once every second. Note that, |volume| is
    // also updated each time SetVolume() is called through IPC by the
    // render-side AGC.
    GetAgcVolume(&normalized_volume);

    while (num_buffers--) {
        int frames_read = wrapper_->PcmReadi(device_handle_, audio_buffer_.get(),
            params_.frames_per_buffer());
        if (frames_read == params_.frames_per_buffer()) {
            audio_bus_->FromInterleaved(audio_buffer_.get(),
                audio_bus_->frames(),
                params_.bits_per_sample() / 8);
            callback_->OnData(
                this, audio_bus_.get(), hardware_delay_bytes, normalized_volume);
        } else {
            LOG(WARNING) << "PcmReadi returning less than expected frames: "
                         << frames_read << " vs. " << params_.frames_per_buffer()
                         << ". Dropping this buffer.";
        }
    }

    next_read_time_ += buffer_duration_;
    base::TimeDelta delay = next_read_time_ - base::TimeTicks::Now();
    if (delay < base::TimeDelta()) {
        DVLOG(1) << "Audio read callback behind schedule by "
                 << (buffer_duration_ - delay).InMicroseconds()
                 << " (us).";
        // Read callback is behind schedule. Assuming there is data pending in
        // the soundcard, invoke the read callback immediate in order to catch up.
        read_callback_behind_schedule_ = true;
        delay = base::TimeDelta();
    }

    base::ThreadTaskRunnerHandle::Get()->PostDelayedTask(
        FROM_HERE,
        base::Bind(&AlsaPcmInputStream::ReadAudio, weak_factory_.GetWeakPtr()),
        delay);
}

void AlsaPcmInputStream::Stop()
{
    if (!device_handle_ || !callback_)
        return;

    StopAgc();

    weak_factory_.InvalidateWeakPtrs(); // Cancel the next scheduled read.
    int error = wrapper_->PcmDrop(device_handle_);
    if (error < 0)
        HandleError("PcmDrop", error);

    callback_ = NULL;
}

void AlsaPcmInputStream::Close()
{
    if (device_handle_) {
        weak_factory_.InvalidateWeakPtrs(); // Cancel the next scheduled read.
        int error = alsa_util::CloseDevice(wrapper_, device_handle_);
        if (error < 0)
            HandleError("PcmClose", error);

        if (mixer_handle_)
            alsa_util::CloseMixer(wrapper_, mixer_handle_, device_name_);

        audio_buffer_.reset();
        device_handle_ = NULL;
        mixer_handle_ = NULL;
        mixer_element_handle_ = NULL;
    }

    audio_manager_->ReleaseInputStream(this);
}

double AlsaPcmInputStream::GetMaxVolume()
{
    if (!mixer_handle_ || !mixer_element_handle_) {
        DLOG(WARNING) << "GetMaxVolume is not supported for " << device_name_;
        return 0.0;
    }

    if (!wrapper_->MixerSelemHasCaptureVolume(mixer_element_handle_)) {
        DLOG(WARNING) << "Unsupported microphone volume for " << device_name_;
        return 0.0;
    }

    long min = 0;
    long max = 0;
    if (wrapper_->MixerSelemGetCaptureVolumeRange(mixer_element_handle_,
            &min,
            &max)) {
        DLOG(WARNING) << "Unsupported max microphone volume for " << device_name_;
        return 0.0;
    }
    DCHECK(min == 0);
    DCHECK(max > 0);

    return static_cast<double>(max);
}

void AlsaPcmInputStream::SetVolume(double volume)
{
    if (!mixer_handle_ || !mixer_element_handle_) {
        DLOG(WARNING) << "SetVolume is not supported for " << device_name_;
        return;
    }

    int error = wrapper_->MixerSelemSetCaptureVolumeAll(
        mixer_element_handle_, static_cast<long>(volume));
    if (error < 0) {
        DLOG(WARNING) << "Unable to set volume for " << device_name_;
    }

    // Update the AGC volume level based on the last setting above. Note that,
    // the volume-level resolution is not infinite and it is therefore not
    // possible to assume that the volume provided as input parameter can be
    // used directly. Instead, a new query to the audio hardware is required.
    // This method does nothing if AGC is disabled.
    UpdateAgcVolume();
}

double AlsaPcmInputStream::GetVolume()
{
    if (!mixer_handle_ || !mixer_element_handle_) {
        DLOG(WARNING) << "GetVolume is not supported for " << device_name_;
        return 0.0;
    }

    long current_volume = 0;
    int error = wrapper_->MixerSelemGetCaptureVolume(
        mixer_element_handle_, static_cast<snd_mixer_selem_channel_id_t>(0),
        &current_volume);
    if (error < 0) {
        DLOG(WARNING) << "Unable to get volume for " << device_name_;
        return 0.0;
    }

    return static_cast<double>(current_volume);
}

bool AlsaPcmInputStream::IsMuted()
{
    return false;
}

void AlsaPcmInputStream::HandleError(const char* method, int error)
{
    LOG(WARNING) << method << ": " << wrapper_->StrError(error);
    callback_->OnError(this);
}

} // namespace media
